[asterisk-users] Hangup hook to put back a call into a queue
hi,
I hope someone can help me:-)
we’ve got a freepbx server. there are 2 special extensions (2001, 2002).
if someone calls this extensions (or a call is forwarded to these
extensions) and these extension hangup (not the caller party), then we’d
like to put the calls back into a queue (1000) and wouldn’t like to hangup.
I read your description about hangup hooks:
https://community.freepbx.org/t/hooking-for-fun-and-income/57718
but still not able to implement it:-(
what I’ve done:
* found out in a hard way how to detect the current destination
extension (because it’s turn out that CALLERID(dnid) is not working in
case of forwarded call it’s show the original destination)
* write a macro-dialout-one-predial-hook and a hook marco like this:
[macro-dialout-one-predial-hook]
exten => s,1,Noop(Entering user defined context
macro-dialout-one-predial-hook in extensions_custom.conf)
exten => s,n,GotoIf($["${DEXTEN}"=“2001”]?special)
exten => s,n,GotoIf($["${DEXTEN}"=“2002”]?special)
exten => s,n,MacroExit
exten => s,n(special),NoOp(--- Push Special Hangup Handler
--)
exten => s,n,Set(CHANNEL(hangup_handler_push)=back-to-1000-hangup,s,1)
exten => s,n,MacroExit
[back-to-1000-hangup]
exten => s,1,Noop(== Entering user defined context
back-to-1000-hangup ===)
exten => s,n,Queue(1000)
exten => s,n,Return
it seems to be called and seem to enter into to call but immediately hangup.
first of all, in this case when in the hangup handler I will NOT like to
hangup how should I finish the marco?:
Hangup
Return
MacroExit
how to redirect the call to the queue?:
Queue(1000)
ChannelRedirect(${CHANNEL},,1000,1)
Gosub(ext-intercom,*801000,1())
dial-one,HhTtrM(auto-blkvm),1000
and what is the reason I can’t put the call back to the queue?
I know that I'm already in the hangup sequence, but still wouldn't like
to hangup.
or this can't be done in the hangup handler?
thank you for your help in advance.
regards.
--
Levente "Si vis pacem para bellum!"
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: https://community.asterisk.org/
New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] how to set 'transfercapability'
hi, we've got a Diva Server BRI-2M PCI, SN:3485 card in our asterisk server. we use the latest divas4linux-melware-3.0.g-106.628.1-1 driver for it. the card is connected to a Bosch Integral33 PBX. the two system connected with an S0 line in order the two pbx be able to call each other. when we call from the bosch to asterisk everything is working properly. but when we call from the a x-ten soft phone client through asterisk to the bosch the it's not working. which means the asterisk pass the call to the bosch, bosch receive but don't ring the given number. after we debug the capi layer with bosch experts from bosch we found the while the bosch call asterisk it request SPEECH line bearer, but when asterisk call bosch it set bearer to MULTIUSE. i found it in divactrl/common/dbg_tapi.c LINE_BEARER_MODE__SPEECH, LINE_BEARER_MODE__MULTIUSE. so probably the problem is thet we (x-ten, asterisk, divas4linux) do not set the bearer (transfercapability) to proper value. is this the real reason? how can i set the bearer/transfercapability to speech in divas4linux or in capi or in asterisk's capi or ...? why the system do not recognize the problem? why x-ten soft phone do not ask for speech mode or why asterisk do not set the transfercapability to speech when it get a call from a soft phone? or what else can we do;-)? thank you for your help in advance. yours. -- Levente Si vis pacem para bellum! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] capi (divas4linux) bearer setting
hi, we've got a Diva Server BRI-2M PCI, SN:3485 card in our asterisk server. we use the latest divas4linux-melware-3.0.g-106.628.1-1 driver for it. the card is connected to a Bosch Integral33 PBX. the two system connected with an S0 line in order the two pbx be able to call eachother. when we call from the bosch to asterisk everything is working properly. but when we call from the a x-ten soft phone client through asterisk to the bosch the it's not working. which means the asterisk pass the call to the bosch, bosch receive but don't ring the given number. after we debug the capi layer with bosch experts from bosch we found the while the bosch call asterisk it request SPEECH time bearer, but when asterisk call bosch it set bearer to MULTIUSE. i found it in ./divactrl/common/dbg_tapi.c LINE_BEARER_MODE__SPEECH, LINE_BEARER_MODE__MULTIUSE. so probably the problem is thet we (x-ten, asterisk, a divas4linux do not set the bearer to proper value. is this the real reason? how can i set the bearer to speech in divas4linux or in capi or in asterisk's capi or ...? thank you for your help in advance. yours. -- Levente Si vis pacem para bellum! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
